HP MSR Router Series Voice Command Reference(V7) Part number: 5998-5634 Software version: CMW710-R0106 Document version: 6PW100-20140607
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Contents Voice interface commands ·········································································································································· 1 Analog voice interface commands ·································································································································· 1 area ·······································································································································································
send-busytone time ················································································································································ 38 shutdown ································································································································································ 38 signal ·····································································································································································
display voice entity················································································································································ 79 entity ······································································································································································· 83 ip qos dscp···························································································································································
reset voice sip connection ·································································································································· 131 session refresh······················································································································································ 131 session transport ·················································································································································· 132 set pstn-ca
Voice interface commands Analog voice interface commands area Use area to specify the standard of busy tones for all the FXO interfaces on the device. Use undo area to restore the default. Syntax area { custom | europe | north-america } undo area Default The European standard is used. Views Voice view Predefined user roles network-admin Parameters custom: Specifies custom busy tones. europe: Specifies the European standard. north-america: Specifies the North American standard.
Parameters index: Assigns a number to a busy tone type, in the range of 0 to 3. The device can record a maximum of four types of busy tones. line-number: Specifies an FXO interface. Usage guidelines The busytone-detect auto command takes effect only on FXO interfaces. After detecting busy tones by using the busytone-detect auto command, the device automatically does the following: • Calculates the busy tone parameters. • Executes the busytone-detect custom command to record the busy tone parameters.
f1: Frequency 1 in Hz, in the range of 50 to 3600. f2: Frequency 2 in Hz, in the range of 50 to 3600. p1: Signal amplitude 1, in the range of 50 to 32767. p2: Signal amplitude 2, in the range of 50 to 32767. p3: Duration of a single tone in milliseconds, in the range of 10 to 1000. p4: Duration error of a single tone in milliseconds, in the range of 0 to 500. p5: Duration of silence in milliseconds, in the range of 10 to 1000. p6: Duration error of silence in milliseconds, in the range of 0 to 500.
Usage guidelines Increasing the number of busy tone detection periods can improve the detection accuracy to reduce the likelihood of false on-hook, but it increases the likelihood of failed on-hook. Test the new value multiple times to make sure the new value does not cause failed on-hook. Examples # Set the number of busy tone detection periods to three.
Use undo calling-name to remove the calling name. Syntax calling-name text undo calling-name Default No calling name is configured. Views FXS interface view Predefined user roles network-admin Parameters text: Specifies a calling name, a case-sensitive string of 1 to 50 characters. Usage guidelines The calling name can be sent only in the multiple-data-message format. Use this command on the originating device. Examples # Configure the calling name as tony for FXS interface 2/1/1.
Parameters 2-wire: Specifies the 2-wire cable type, which provides full-duplex voice transmission. Each wire carries bidirectional signals. 4-wire: Specifies the 4-wire cable type, which provides simplex voice transmission. Every two wires receive and send signals in one direction. Usage guidelines You must configure the same cable type for the E&M interfaces on the originating and terminating devices. Otherwise, only one-way voice communication can be implemented.
Syntax cid receive undo cid receive Default CID receiving is enabled for an FXO interface. Views FXO interface view Predefined user roles network-admin Usage guidelines CID receiving must be enabled for the CID function to work correctly. Examples # Enable CID receiving on FXO interface 2/2/1.
Examples # Configure FXO interface 2/2/1 to detect CID before the rings. system-view [Sysname] subscriber-line 2/2/1 [Sysname-subscriber-line2/2/1] cid ring 0 cid send Use cid send to enable CID sending on an FXS or FXO interface. Use undo cid send to disable CID sending. Syntax cid send undo cid send Default CID sending is enabled on an FXS or FXO interface.
Parameters complex: Specifies MDMF. simple: Specifies single data message format (SDMF). Usage guidelines The local and remote ends must use the same CID format. The calling name in the CID can only be transmitted in MDMF format. This command takes effect only on the terminating device. Examples # Set the CID format to SDMF on FXS interface 2/1/1.
The CID format configured by using the cid type command takes effect only when the bellcore standard is used. Examples # Specify the CID standard as brazil for FXS interface 2/1/1. system-view [Sysname] subscriber-line 2/1/1 [Sysname-subscriber-line2/1/1] cid standard-type brazil Related commands cid type cptone Use cptone to specify the call progress tones of a country or region. Use undo cptone to restore the default.
Code Country/region name CY Cyprus CZ Czech Republic DK Denmark EG Egypt FI Finland FR France DE Germany GH Ghana GR Greece HK Hong Kong China HU Hungary IS Iceland IN India ID Indonesia IR Iran IE Ireland IEU Ireland (UK style) IL Israel IT Italy JP Japan JO Jordan KE Kenya KR Republic of Korea LB Lebanon LU Luxembourg MO Macau MY Malaysia MX Mexico NP Nepal NL Netherlands NZ New Zealand NG Nigeria NO Norway PK Pakistan PA Panama PH P
Code Country/region name PL Poland PT Portugal RU Russian Federation SA Saudi Arabia SG Singapore SK Slovakia SI Slovenia ZA South Africa ES Spain SE Sweden CH Switzerland TH Thailand TR Turkey GB United Kingdom US United States UY Uruguay ZW Zimbabwe custom: Customizes call progress tones. busy-tone: Specifies the busy tone. congestion-tone: Specifies the congestion tone. dial-tone: Specifies the dial tone. ringback-tone: Specifies the ringback tone.
time4: Specifies the break time for the second make-to-break ratio in milliseconds. The value range is 0 and 30 to 8191. If time1 is set to 0, this argument must be set to 0. Usage guidelines This command takes effect only for the progress tones on the local device. Examples # Specify the call progress tones of US. system-view [sysname] voice-setup [sysname-voice] cptone country-type us # Customize the call progress tones.
Examples # Set the amplitude of the busy tone to 1200. system-view [sysname] voice-setup [sysname-voice] cptone tone-type busy-tone amplitude 1200 cng-on Use cng-on to enable the comfortable noise generation (CNG) function on a voice interface. Use undo cng-on to disable this function. Syntax cng-on undo cng-on Default The CNG function is enabled.
This command might fail to restore the default settings for some commands for reasons such as command dependencies or system restrictions. Use the display this command in interface view to identify these commands. Then use their undo forms or follow the command reference to restore their default settings. If your restoration attempt still fails, follow the error message instructions to resolve the problem. Examples # Restore the default settings for FXS interface 2/1/1.
undo delay rising Default The delay time is 300 milliseconds. Views E&M interface view Predefined user roles 2: System level Parameters rising milliseconds: Specifies the delay time from when the terminating side detects a seizure signal to when it sends the seizure signal in the delay start mode. The value range is 20 to 2000 milliseconds.
[Sysname] subscriber-line 2/3/1 [Sysname-subscriber-line2/3/1] signal immediate [Sysname-subscriber-line2/3/1] delay send-dtmf 3000 Related commands signal delay send-wink Use delay send-wink to configure a delay from when the terminating side receives a seizure signal to when it sends a wink signal in wink start mode. Use undo delay send-wink to restore the default.
Default The dial delay time is 1 second. Views FXS interface view, FXO interface view Predefined user roles network-admin Parameters seconds: Specifies the dial delay time in the range of 0 to 10 seconds. Examples # Set the dial delay time to 5 seconds for FXS interface 2/1/1.
delay wink-rising Use delay wink-rising to configure the timeout time for the originating side to wait for a wink signal after sending a seizure signal in wink start mode. Use undo delay wink-rising to restore the default. Syntax delay wink-rising milliseconds undo delay wink-rising Default The timeout time for the originating side to wait for a wink signal after sending a seizure signal is 2000 milliseconds in wink start mode.
Parameters text: Specifies a description, a case-sensitive string of 1 to 80 characters. Examples # Configure the description as pstn for FXS interface 2/1/1. system-view [Sysname] subscriber-line 2/1/1 [Sysname-Subscriber-line2/1/1] description pstn disconnect lcfo Use disconnect lcfo to configure an FXS interface to send a LCFO signal when the peer goes on-hook. Use undo disconnect lcfo to restore the default.
Predefined user roles network-admin network-operator Parameters line-number: Specifies a voice interface by its number. Examples # Display information about FXS interface 2/1/1. display voice subscriber-line 2/3/1 Current information: subscriber-line2/3/1 Type: E&M Status: Up Call status: Idle Table 2 Command output Field Description Type • FXS. • FXO. • E&M. Status • Down. • Up. • Down(Administratively)—The voice interface is shut down by using the shutdown command.
Field Description • For FXS interfaces: { Idle. { Receiving number. { Ringing. { Listening to ringback tone. { Playing busytone. { Talking. { Releasing. • For FXO interfaces: Call Status { Idle. { Receiving number. { Ringing. { Listening to ringback tone. { Playing busytone { Talking. { Releasing. { Bound and off-hook. { Bound and on-hook. • For E&M interfaces: { Idle. { Sending number. { Ringing. { Listening to ringback tone. { Playing busytone. { Talking.
Examples # Configure the amplitude of DTMF tones as –8.0 dBm. system-view [Sysname] voice-setup [Sysname-voice] dtmf amplitude -8.0 dtmf sensitivity-level Use dtmf sensitivity-level to set the DTMF detection sensitivity level. Use undo dtmf sensitivity-level to restore the default. Syntax dtmf sensitivity-level { high | low | medium [ frequency-tolerance value ] } undo dtmf sensitivity-level Default The DTMF detection sensitivity level is low.
Syntax dtmf time { interval | persist } milliseconds undo dtmf time { interval | persist } Default The duration of DTMF tones and the interval between DTMF tones are both 120 milliseconds. Views Voice view Predefined user roles network-admin Parameters persist: Specifies the duration of DTMF tones. interval: Specifies the interval between DTMF tones. milliseconds: Specifies the time value in the range of 50 to 500 milliseconds.
The maximum energy of the input signal in the row frequency group is ROWMAX, and the corresponding doubled frequency energy is ROW2nd. The maximum energy in the column frequency group is COLMAX, and the corresponding doubled frequency energy is COL2nd. Table 3 Meaning of index numbers Index Meaning Value range Remarks 0 Lower limit of (ROWMAX + COLMAX). The input signal is recognized as a DTMF digit if (ROWMAX + COLMAX) > Value for index 0.
Index Meaning Value range Remarks 9 Lower limit of the DTMF tone duration. The duration of DTMF key tone must be larger than this threshold for the input signal to be recognized as a DTMF digit. 30 to 150 milliseconds, with a default of 30 milliseconds The larger the value, the higher the detection specificity and the lower the detection sensitivity.
Views Voice view Predefined user roles network-admin Parameters convergence-rate value: Sets the convergence rate of comfort noise amplitude, in the range of 0 to 511. The greater the value, the quicker the convergence. max-amplitude value: Sets the maximum amplitude of comfort noise, in the range of 0 to 2048. The greater the value, the greater the noise amplitude. The value 0 indicates that the system performs only nonlinear processing and does not add comfort noise.
Parameters milliseconds: Specifies the echo cancellation delay in the range of 0 to 64 milliseconds. Usage guidelines The echo cancellation delay is the time from when a subscriber speaks to when the subscriber hears the echo. Examples # Enable echo cancellation, and configure the echo cancellation delay as 24 milliseconds on FXS interface 2/1/1.
Use undo echo-canceler tail-length to restore the default. Syntax echo-canceler tail-length milliseconds undo echo-canceler tail-length Default The echo cancellation coverage is 128 milliseconds. Views FXS interface view, FXO interface view, E&M interface view Predefined user roles network-admin Parameters milliseconds: Specifies the echo cancellation coverage in milliseconds. For routers installed with the interface module SIC VE1 or SIC VT1, the value range is 32, 48, 64, 80, 96, 112, and 128.
Predefined user roles network-admin Parameters delay: Specifies the delay off-hook mode. immediate: Specifies the immediate off-hook mode. Examples # Specify the delay off-hook mode for FXO interface 2/2/1. system-view [Sysname] subscriber-line 2/2/1 [Sysname-subscriber-line 2/2/1] hookoff-mode delay hookoff-mode delay bind Use hookoff-mode delay bind to bind an FXS interface to an FXO interface. Use undo hookoff-mode delay bind to remove the binding.
hookoff-time Use hookoff-time to configure forced on-hook. Use undo hookoff-time to disable forced on-hook. Syntax hookoff-time time undo hookoff-time Default Forced on-hook is disabled. Views FXO interface view Predefined user roles network-admin Parameters time: Specifies the time from off-hook to forced on-hook, in the range of 60 to 36000 seconds. Usage guidelines In some countries, PBXs do not play busy tones, or the busy tones only last for a short period of time.
Views FXO interface view, FXS interface view Predefined user roles network-admin Parameters country-name: Specifies a country. It can be Australia, Austria, Belgium-Long, Belgium-Short, Brazil, China, Czech-Republic, Denmark, ETSI-Harmonized, Finland, France, German-Swiss, Greece, Hungary, India, Italy, Japan, Korea, Mexico, Netherlands, New Zealand, Norway, Portugal, Slovakia, Spain, Sweden, U.K., US-Loaded-Line, US-Non-Loaded, or US-Special-Service. r550: 550-ohm real impedance.
Predefined user roles network-admin Usage guidelines This command takes effect only after the echo-canceler enable command is configured. The undo nlp-on command is supported only on router models installed with cards SIC-2FXS1FXO, HMIM-8FXS8FXO, DSIC-4FXS1FXO, or HMIM-16FXS. Examples # Disable the EC nonlinear processing function on FXO interface 2/2/1.
For the E&M non-signaling mode to work with the PLAR function, you must configure the private-line command on the originating device. For more information about the PLAR function, see "Configuring dial programs." Examples # Enable the E&M non-signaling mode for E&M interface 2/3/1 on the originating device, and specify the monitoring time as 120 seconds.
Default The gateway-specific compensation mode is used. Views FXS interface view, FXO interface view Predefined user roles network-admin Parameters general: Uses the general compensation mode to reconstruct lost packets. This mode applies to discrete packet loss. specific: Uses the voice gateway-specific compensation mode to reconstruct lost packets. This mode applies to continuous packet loss. Examples # Configure the general compensation mode to reconstruct lost packets on FXS interface 2/1/1.
Related commands transmit gain ring-detect debounce Use ring-detect debounce to configure the debounce time for ring detection. Use undo ring-detect debounce to restore the default. Syntax ring-detect debounce value undo ring-detect debounce Default The debounce time is 10 milliseconds. Views FXO interface view Predefined user roles network-admin Parameters value: Specifies the debounce time for ring detection, in the range of 4 to 15 milliseconds.
Default The frequency for ring detection is 40 Hz. Views FXO interface view Predefined user roles network-admin Parameters value: Specifies the frequency value for ring detection, in Hz. The value is in the range of 30 to 100 in increments of 10. Usage guidelines This command is supported only on routers installed with the interface module SIC-2FXS1FXO, HMIM-8FXS8FXO, or DSIC-4FXS1FXO. Examples # Set the frequency value for ring detection on FXO interface 2/2/1 to 100 Hz.
send-busytone time Use send-busytone time to configure the busy tone duration. Use undo send-busytone time to restore the default. Syntax send-busytone time seconds undo send-busytone time Default The busy tone duration is 3 seconds. Views FXO interface view Predefined user roles network-admin Parameters time seconds: Specifies the busy tone duration in the range of 2 to 15 seconds. Usage guidelines The send-busytone time command takes effect only after you configure the send-busytone enable command.
Predefined user roles network-admin Examples # Shut down FXS interface 2/1/1. system-view [Sysname] subscriber-line 2/1/1 [Sysname-subscriber-line2/1/1] shutdown signal Use signal to configure a start mode for an E&M interface. Use undo signal to restore the default. Syntax signal { delay | immediate | wink } undo signal Default An E&M interface uses the immediate start mode. Views E&M interface view Predefined user roles network-admin Parameters delay: Specifies delay start mode.
Default The silence threshold is 20, and the silence duration for automatic on-hook is 7200 seconds (2 hours). Views FXO interface view Predefined user roles network-admin Parameters threshold: Specifies the silence threshold in the range of 0 to 200. If the amplitude of voice signals from the PBX is smaller than this value, the system regards the voice signals as silence. time-length: Specifies the silence duration for automatic on-hook, in the range of 2 to 7200 seconds.
1: Sets the output gain of the SLIC chip to 2.1 dB. Examples # Set the output gain of the SLIC chip to 1 (2.1 dB) for E&M interface 2/3/1. system-view [Sysname] subscriber-line 2/3/1 [Sysname-subscriber-line2/3/1] slic-gain 1 subscriber-line Use subscriber-line to enter voice interface view. Syntax subscriber-line line-number Views System view Predefined user roles network-admin Parameters line-number: Voice interface number. Examples # Enter the view of FXS interface 2/1/1.
Usage guidelines This timer restarts each time the subscriber dials a digit. If the timer expires before the subscriber dials the next digit, the system prompts the subscriber that the dialing times out. The maximum interval from off-hook to dialing the first digit is set by the timer first-dial command. Examples # Configure the maximum interval for dialing the next digit to 5 seconds on FXS interface 2/1/1.
Default The timeout time between off-hook and dialing the first digit is 10 seconds. Views FXS interface view, FXO interface view Predefined user roles network-admin Parameters seconds: Specifies the timeout time between off-hook and dialing the first digit, in the range of 1 to 300 seconds. Usage guidelines If the timer expires before the subscriber dials the first digit, the system prompts the subscriber that the dialing times out.
timer hookoff-interval Use timer hookoff-interval to configure the interval between on-hook and off-hook. Use undo timer hookoff-interval to restore the default. Syntax timer hookoff-interval milliseconds undo timer hookoff-interval Default The interval between on-hook and off-hook is 500 milliseconds. Views FXO interface view Predefined user roles network-admin Parameters milliseconds: Specifies the interval between on-hook and off-hook, in the range of 500 to 4000 milliseconds.
Predefined user roles network-admin Parameters seconds: Specifies the maximum duration for playing ringback tones, in the range of 5 to 120 seconds. Examples # Configure the maximum duration for playing ringback tones as 8 seconds for FXS interface 2/1/1. system-view [Sysname] subscriber-line 2/1/1 [Sysname-subscriber-line2/1/1] timer ring-back 8 timer wait-digit Use timer wait-digit to configure the timeout time for the terminating device to wait for the first digit.
Syntax transmit gain value undo transmit gain Default The output gain value is 0 dB. Views FXS interface view, FXO interface view, E&M interface view Predefined user roles network-admin Parameters value: Specifies the output gain value in the range of –14.0 to +13.9 dB. Usage guidelines If the power of output voice signals is larger than the power required by the output line, you can use this command to reduce the output gain. Output gain adjustment might lead to call failures.
3: Specifies E&M signal type III. 5: Specifies E&M signal type V. Usage guidelines You must configure the same E&M signal type on the originating and terminating devices. Examples # Configure the signal type as 3 for E&M interface 2/3/1. system-view [Sysname] subscriber-line 2/3/1 [Sysname-subscriber-line2/3/1] type 3 Digital voice interface commands ani Use ani to configure the terminating side to request calling information (calling category and calling number) from the originating side.
ani-digit Use ani-digit to set the number of dialed digits that the terminating side collects before requesting calling information. Use undo ani-digit to restore the default. Syntax ani-offset number undo ani-offset Default The number of dialed digits that the terminating side collects before requesting calling information is 1. Views R2 CAS view Predefined user roles network-admin Parameters number: Specifies the number of dialed digits, in the range of 1 to 10.
Default The originating side requires the terminating side to send answer signals. Views R2 CAS view Predefined user roles network-admin Usage guidelines If the originating side does not require the terminating side to send answer signals, it directly establishes a call with the terminating side. Otherwise, the originating side establishes a call with the terminating side after receiving answer signals. Examples # Configure the originating side to not require the terminating side to send answer signals.
cas Use cas to enter R2 CAS view. Use undo cas to exit R2 CAS view and delete the settings in R2 CAS view. Syntax cas ts-set-number undo cas ts-set-number Views E1 interface view, T1 interface view Predefined user roles network-admin Parameters ts-set-number: Specifies a timeslot set by its number in the range of 0 to 30 for an E1 interface, or in the range of 0 to 23 for a T1 interface.
Predefined user roles network-admin Examples # Configure the terminating side to send a clear-back signal when the originating side first disconnects the line. system-view [Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 2/4/1] cas 0 [Sysname-cas 2/4/1:0] clear-forward-ack enable display voice subscriber-line Use display voice subscriber-line to display information about digital voice interfaces.
TS 4: Idle TS 5: Idle TS 6: Idle TS 7: Idle TS 8: Idle TS 9: Idle TS 10: Idle TS 11: Idle TS 12: Idle TS 13: Idle TS 14: Idle TS 15: Idle TS 17: Idle TS 18: Idle # Display information about voice interface 2/5/1 generated on a BSV interface. display voice subscriber-line 2/5/1 Current information : subscriber-line2/5/1 Type: ISDN Status: Up Call status: TS 0: Idle TS 1: Idle # Display information about a subinterface of voice interface 2/5/1.1 generated on a BSV interface.
Field Description • For R2: Call Status { Idle. { Seize. { Seize Ack. { Talking. { Releasing. • For ISDN: { Idle { Call in. { Call out. { Ring. { Ringback tone. { Talking. { Releasing. dl-bits Use dl-bits to configure the ABCD bit pattern for line signals. Use undo dl-bits to restore the default.
Predefined user roles network-admin Parameters answer: Specifies the answer signal. blocking: Specifies the blocking signal. clear-back: Specifies the clear-back signal. clear-forward: Specifies the clear-forward signal. idle: Specifies the idle signal. seizing: Specifies the seizure signal. seizing-ack: Specifies the seizure acknowledgement signal. release-guard: Specifies the release guard signal. receive: Applies the signaling setting to received line signals.
• MFC—The originating and terminating sides use interregister signaling to transmit and request number information, including the calling number, line information, and billing. In the exchange process, the terminating side sends responses to the originating side. • DTMF—The originating side transmits the called number to the terminating side digit by digit. The terminating side does not send any responses for confirmation.
[Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 2/4/1] cas 0 [Sysname-cas 2/4/1:0] final-callednum enable group-b enable Use group-b enable to configure R2 signaling to use Group B signals to complete registers exchange. Use undo group-b enable to configure R2 signaling to not use Group B signals to complete registers exchange. Syntax group-b enable undo group-b enable Default R2 signaling uses Group B signals to complete registers exchange.
Predefined user roles network-admin Parameters line-number: Specifies an E1 or T1 interface by its number. ts-set-number: Specifies a timeslot set by its number. 15: Number for the PRI set created by bundling the timeslots of an E1 interface. 23: Number for the PRI set created by bundling the timeslots of a T1 interface. Examples # Bind a digital voice interface to POTS voice entity 10.
mode Use mode to specify the R2 signaling standard. Use undo mode to restore the default. Syntax mode zone-name [ default-standard ] undo mode Default ITU-T R2 signaling is used. Views R2 CAS view Predefined user roles network-admin Parameters zone-name: Specifies a country or region from the following list: • argentina: Argentina. • australia: Australia. • bengal: Bengal. • brazil: Brazil. • china: China. • custom: Custom. • hongkong: Hong Kong. • india: India. • indonesia: Indonesia.
system-view [Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 2/4/1] cas 0 [Sysname-cas 2/4/1:0] mode singapore pcm Use pcm to configure a companding law for PCM. Use undo pcm to restore the default. Syntax pcm { a-law | μ-law } undo pcm Default The companding law for PCM is a-law for E1 interfaces and μ-law for T1 interfaces.
Views R2 CAS view Predefined user roles network-admin Usage guidelines R2 signaling in some countries must support re-answer processing. When the terminating side sends a clear-back signal, the originating side does not release the line, but maintains the call state. If it receives a re-nswer signal from the terminating side within a specified time, it continues the call. Otherwise, it disconnects the call upon timeout. Examples # Enable reanswer signal processing on the originating side.
Parameters billingcategory value: Specifies the billing category value in the range of 1 to 16. It configures the KA signal in R2 signaling. The signal provides two types of information for a call connection: billing category (regular, immediate, or toll free) and subscriber level (with or without priority). callcreate-in-groupa value: Specifies the direct call setup signal value in the range of 1 to 16. callingcategory value: Specifies the calling category signal value in the range of 1 to 16.
Usage guidelines The register-value command assigns values to signals requesting responses from the remote end. For example, after you configure the register-value callingcategory command, the terminating side sends the calling category signal with the specified value to the originating side for the calling category. A signal value of 16 disables the signal function. HP recommends that you use the default values.
[Sysname-cas 2/4/1:0] renew 0011 Related commands mode reverse Use reverse to enable line signal inversion. Use undo reverse to restore the default. Default Line signal inversion is disabled (ABCD takes the value of 0000). Syntax reverse ABCD undo reverse Views R2 CAS view Predefined user roles network-admin Parameters ABCD: Indicates whether corresponding ABCD bits in R2 signaling need inversion. Each argument in this command takes 0 or 1.
Views R2 CAS view Predefined user roles network-admin Parameters max: Selects the timeslot with the greatest number from available timeslots. maxpoll: Selects the timeslot with the greatest number from available timeslots in the first timeslot polling. Subsequent pollings select in descending order timeslots with numbers less than the one selected in the previous polling. For example, suppose timeslots 31 and 29 are not available.
Examples # Configure the originating side to not require the terminating side to send seizure acknowledgement signals. system-view [Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 2/4/1] cas 0 [Sysname-cas 2/4/1:0] undo seizure-ack enable send ringbusy enable Use send ringbusy enable to configure the terminating side to send busy tones to the originating side.
undo special-character character number Default No signal code is configured for a special character. Views R2 CAS view Predefined user roles network-admin Parameters character: Specifies a special character, which can be a pound sign (#), asterisk (*), A, B, C, or D. number: Specifies a signal code in the range of 11 to 15. Usage guidelines R2 signaling in some countries includes special characters such as pound signs (#) and asterisks (*) in Group I forward signals.
After you create a PRI group with the pri-set command on an E1/T1 interface, the system automatically creates a voice interface numbered E1 interface-number:15 on an E1 interface and T1 interface-number:23 on a T1 interface. Examples # Enter the view of voice interface 2/4/1:15. system-view [Sysname] subscriber-line 2/4/1:15 [Sysname-subscriber-line2/4/1:15] Related commands • timeslot-set • pri-set tdm-clock Use tdm-clock to configure a TDM clock source for an E1/T1 interface.
• If the line keyword is specified for one interface and the internal keyword for all other interfaces, the clock on that one interface is used. • The clock source of only one interface can be set to line primary. The TDM clock sources on the local and peer devices must match. For example, if the clock source is set to line for a subsystem on the local device, the clock source must be set to internal on the peer device, and vice versa.
[Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 0 timeslot-list 1-31 signal r2 [Sysname-E1 2/4/1] cas 0 [Sysname-cas 2/4/1:0] timer ringback 10000 timer dl Use timer dl to set the timeout time of line signals. Use undo timer dl to restore the default.
release-guard time: Specifies the timeout time of release guard signals, in the range of 100 to 60000 milliseconds. The originating side starts this timer after sending a clear-forward signal. If it does not receive a release guard signal from the terminating side before the timer expires, it clears the connection. This option applies to the originating side. seizing time: Specifies the timeout time of seizure signals, in the range of 100 to 5000 milliseconds.
[Sysname-cas 2/4/1:0] timer dtmf-delay 800 Related commands dtmf enable timer group-b Use timer group-b to configure the timeout time for Group B signal exchange. Use undo timer group-b to restore the default. Syntax timer group-b time undo timer group-b Default The timeout time is 30000 milliseconds. Views R2 CAS view Predefined user roles network-admin Parameters group-b time: Specifies the timeout time for Group B signal exchange, in the range of 100 to 90000 milliseconds.
Default The duration is 150 milliseconds. Views R2 CAS view Predefined user roles network-admin Parameters time: Specifies the duration of register pulse signals, in the range of 50 to 3000 milliseconds. Examples # Configure the duration of register pulse signals as 300 milliseconds.
Examples # Create timeslot set 5, which contains timeslots 1 through 31 and uses R2 signaling. system-view [Sysname] controller e1 2/4/1 [Sysname-E1 2/4/1] timeslot-set 5 timeslot-list 1-31 signal r2 trunk-direction Use trunk-direction to configure the trunk direction. Use undo trunk-direction to restore the default. Syntax trunk-direction timeslots timeslots-list { dual | in | out } undo trunk-direction timeslots timeslots-list Default Bidirectional trunking applies.
ts Use ts to maintain specified timeslots. Syntax ts { block | open | query | reset } timeslots timeslots-list Views R2 signaling view Predefined user roles network-admin Parameters block: Blocks the specified timeslots to make them unavailable. open: Opens the specified timeslots to make them available. query: Queries the status of the specified timeslots to see whether they are busy, open, or blocked. reset: Resets the specified timeslots.
Voice entity commands codec Use codec to configure a codec for a voice entity. Use undo codec to delete the configured codec. Syntax codec { g711alaw | g711ulaw | g723r53 | g723r63 | g726r16 | g726r24 | g726r32 | g726r40 | g729a | g729br8 | g729r8 } [ bytes payload-size ] undo codec Default No codecs are configured.
Table 6 Value range and default of payload-size for codecs Codec Value range (in bytes) Default (in bytes) 80 to 240 in multiples of 80 160 g723r53 20 to 120 in multiples of 20 20 g723r63 24 to 144 in multiples of 24 24 g726r16 20 to 220 in multiples of 20 60 g726r24 30 to 210 in multiples of 30 90 g726r32 40 to 200 in multiples of 40 120 g726r40 50 to 200 in multiples of 50 150 10 to 180 in multiples of 10 30 g711alaw g711ulaw g729r8 g729br8 g729r8 Usage guidelines If you configu
Examples # Configure the codec as g711alaw for VoIP entity 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 voip [Sysname-voice-dial-entity10] codec g711alaw codec preference Use codec preference to assign a priority to a codec in a codec template. Use the undo codec preference command to delete the assigned priority.
undo description Default No description is configured. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters string: Specifies a description, a case-sensitive string of 1 to 80 characters. Examples # Configure the description as room10 for POTS entity 10.
Table 8 Command output Field Description Call direction • From packet switch—The call is initiated from the IP side. • From circuit switch—The call is initiated from the PSTN side. Voice interface index Index of the voice interface that initiates the call. Voice entities offered Number of voice entities that can be used for the call. display voice entity Use display voice entity to display the configuration of voice entities.
Jitter-buffer minimum delay: 40 ms Jitter-buffer maximum delay: 200 ms IP media DSCP: ef Register number: Enabled Voice class codec: 1 Call-forwarding no-reply number: 123 Call-forwarding on-busy number: 111 Call-forwarding unavailable number: 11111 Call-forwarding unconditional number: 1234 Fax protocol: standard-t38; ls-redundancy: 0; hs-redundancy: 0 Fax cng-switch: Disabled Fax level: -15 Fax local-train threshold: 10 Fax nsf: 0x000000 Fax rate: Voice Fax train-mode: PPP Fax ecm: Disabled Authentication
IP media DSCP: ef Codec transparent: Disabled Media flow-around: Disabled Media protocol: Global Voice class SIP early-offer forced: Disabled Voice class SIP URI scheme: Global Voice class SIP bind media source-interface: GigabitEthernet2/1/1 Voice class SIP bind control source-interface: GigabitEthernet2/1/1 Voice class SIP keepalive up-interval: 60 s Voice class SIP keepalive down-interval: 30 s Voice class SIP keepalive retry: 5 Fax protocol: standard-t38; ls-redundancy: 0; hs-redundancy: 0 Fax cng-switc
Field Description Jitter-buffer initial delay This field is not supported and is reserved for future support. Jitter-buffer minimum delay This field is not supported and is reserved for future support. Jitter-buffer maximum delay This field is not supported and is reserved for future support. IP media DSCP DSCP value of IP packets carrying streaming media. Codec transparent State of transparent transmission of codecs: Enabled or Disabled.
entity Use entity to create a voice entity and enter voice entity view. Use undo entity to delete an existing voice entity. Syntax entity entity-number [ pots | voip ] undo entity { entity-number | all | pots | voip } Default No voice entities exist. Views Voice dial program view Predefined user roles network-admin Parameters entity-number: Number of the voice entity to be created, in the range of 1 to 2147483647. all: Specifies all voice entities. pots: Specifies the POTS entity.
Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters dscp-value: Specifies a DSCP value in the range of 0 to 63. dscp-value-set: DSCP value, which can be the keyword af11, af12, af13, af21, af22, af23, af31, af32, af33, af41, af42, af43, cs1, cs2, cs3, cs4, cs5, cs6, cs7, or ef.
system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] ip qos dscp af41 media Related commands ip qos dscp (in SIP view) line Use line to bind a voice interface to a POTS entity. Use undo line to remove the binding. Syntax line line-number undo line Default There is no binding between a voice interface and a POTS entity.
Views POTS entity view, VoIP entity view Predefined user roles 2: System level Parameters match-string: Specifies a number template, a string of 1 to 31 characters that is in the format of [ + ] { string [ T ] [ $ ] | T }. The following describe the characters: Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.164 standard number. For example, +110022 indicates that 110022 is an E.164 standard number.
Character Description Plus sign (+) Indicates the sub-expression before it appears one or more times. For example, 9876(54)+ can match 987654, 98765454, 9876545454, and so on. Percent sign (%) Indicates the sub-expression before it appears multiple times or does not appear. For example, 9876(54)% can match 9876, 987654, 98765454, 9876545454, and so on. Hyphen (-) the sign is similar to that of the wildcard dot (.). These signs must follow a valid digit or digit string.
Syntax outband nte undo outband Default Inband DTMF is used. Views POTS entity view, VoIP entity view, Predefined user roles network-admin Usage guidelines HP recommends that you configure the outband nte command and the same payload type value on the originating and terminating devices. Otherwise, DTMF tones might fail to be transmitted. Examples # Enable NTE mode for out-of-band DTMF.
When the device is connected to a device from another vendor, you cannot set the payload type field to any value forbidden by that device. Otherwise, NTE negotiation might fail. Examples # Set the NTE payload type field to 102 for VoIP entity 10.
Default VAD is disabled. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters g723r53: Enables VAD for the G723.1 Annex A codec at 5.3 kbps. g723r63: Enables VAD for the G723.1 Annex A codec at 6.3 kbps. g729a: Enables VAD for the G729 Annex A codec at 8 kbps. g729r8: Enables VAD for the G729 codec at 8 kbps. Usage guidelines If you execute the vad-on or undo vad-on command without specifying a codec, VAD for all codecs is enabled or disabled. The G.711 and G.
Usage guidelines The device supports a maximum of 16 codec templates. Examples # Create codec template 1. system-view [Sysname] voice-setup [Sysname-voice] voice class codec 1 [sysname-voice-class-codec1] voice-class codec Use voice-class codec to bind a codec template to a voice entity. Use undo voice-class codec to delete the binding. Syntax voice-class codec tag undo voice-class codec tag Default The binding does not exist.
voice-setup Use voice-setup to enter voice view and enable voice services. Use undo voice-setup to disable voice services, delete all voice settings, and exit voice view. Syntax voice-setup undo voice-setup Default Voice services are disabled. Views Voice view Predefined user roles network-admin Examples # Enter voice view and enable voice services.
Dial program commands caller-group Use caller-group to configure a voice entity to permit or deny the calling numbers in a subscriber group. Use undo caller-group to remove the configuration. Syntax caller-group { deny | permit } group-id undo caller-group { { deny | permit } group-id | all } Default A voice entity permits all calling numbers. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters deny: Denies the calling numbers in the subscriber group.
Default A voice entity permits all calling numbers. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters all: Removes all the configured calling numbers. calling-string: Specifies a string of 1 to 31 characters in the format of { [ + ] string [ $ ] }| $. The voice entity uses the string to match calling numbers. The following describes the symbols in the format: • Plus sign (+): If the plus sign (+) is at the beginning of the string, the string indicates an E.
Character Description Hyphen (-) Used to connect two digits to indicate a range of numbers, for example, [1-9] indicates 1 to 9 inclusive. The hyphen (-) can present only in brackets ([ ]). Brackets ([ ]) Indicates a range. For example, [1-36A] matches 1, 2, 3, 6, or A. Parentheses (( )) Indicates a string of characters. For example, (123) indicates the character string 123. It is usually used together with signs such as !, %, and +.
Examples # Configure a description of international for subscriber group 10. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] subscriber-group 10 [Sysname-voice-dial-group10] description international dial-prefix Use dial-prefix to configure a dial prefix for a POTS entity. Use undo dial-prefix to remove the dial prefix of a POTS entity. Syntax dial-prefix string undo dial-prefix Default No dial prefix is configured for a voice entity.
Related commands match-template dial-program Use dial-program to enter dial program view. Use undo dial-program to remove all the settings from dial program view. Syntax dial-program undo dial-program Views Voice view Predefined user roles network-admin Examples # Enter dial program view. system-view [Sysname] voice-setup [Sysname-voice] dial-program dot-match Use dot-match to configure a dot match rule. Use undo dot-match to restore the default.
NOTE: The input and output templates are configured using the rule command. Examples # Set the dot match rule to right-left for number substitution rule list 20. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 20 [Sysname-voice-dial-substitute20] dot-match right-left Related commands rule entity hunt Use entity hunt to configure a voice entity selection order. Use undo entity hunt to restore the default.
Table 14 Selection rules Selection rule Description Longest match Selects the voice entity that matches (from left to right) the most digits of the dialed number. Voice entity priority Selects the voice entity with the highest priority (configured by using the priority command). Random selection Selects a voice entity in a random manner. Least recent use Selects the voice entity that has waited for the longest time since being last selected.
Examples # Specify rule 4 in number substitution list 20 as the preferred number substitution rule. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 20 [Sysname-voice-dial-substitute20] rule 4 663 3 [Sysname-voice-dial-substitute20] first-rule 4 Related commands rule match-template Use match-template to configure a calling number match template for a subscriber group.
Table 15 Description of characters in a string Character Description 0-9 Digits 0 through 9. Pound sign (#) or asterisk (*) Indicates a valid digit. Dot (.) Wildcard, which can match any valid digit. For example, 555…. can match any 7-digit number beginning with 555. Exclamation point (!) Indicates the subexpression before it appears once or does not appear. For example, 56!1234 can match 51234 and 561234. Plus sign (+) Indicates the subexpression before it appears one or more times.
Use undo max-conn to restore the default. Syntax max-conn max-number undo max-conn Default No maximum number of total calls is configured. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters max-number: Specifies the maximum number of total calls allowed by a voice entity, in the range of 0 to 120. Examples # Configure the maximum number of total calls allowed by VoIP entity 10 as 5.
system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] number-match longest Related commands terminator number-substitute Use number-substitute to create a number substitution rule list and enter number-substitute view. Use undo number-substitute to remove a number substitution rule list or all number substitution rule lists. Syntax number-substitute list-number undo number-substitute { list-number | all } Default No number substitution rule list is configured.
Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters priority-order: Specifies a priority in the range of 0 to 10. The smaller the value, the higher the priority. Usage guidelines If a number matches multiple voice entities, the router selects the voice entity with the highest priority. Examples # Set the priority to 5 for POTS entity 10.
rule Use rule to configure a number substitution rule. Use undo rule to remove a number substitution rule or all number substitution rules. Syntax rule id input-template output-template [ number-type input-number-type output-number-type | numbering-plan input-numbering-plan output-numbering-plan ] * undo rule { id | all } Default No number substitution rule is configured. Views Number-substitute view Predefined user roles network-admin Parameters all: Deletes all number substitution rules.
output-template: Configures an output template, which is a string of 1 to 31 characters that can include digits 0 through 9, pound sign (#), asterisk (*), plus sign (+) and dot (.). The first character can be a plus sign (+). Table 16 describes these characters. number-type: Specifies input and output number types. input-template-type: Specifies an input number type. Table 17 describes the values for this argument. Table 17 Input number types Number type Description abbreviated Abbreviated number.
Numbering plan Description reserved Reserved numbering plan. telex Telex numbering plan. unknown Unknown numbering plan. output-numbering-plan: Specifies an output numbering plan. Table 20 describes the values for this argument. Table 20 Output numbering plans Numbering plan Description data Data numbering plan. isdn ISDN telephone numbering plan. national National numbering plan. private Private numbering plan. reserved Reserved numbering plan. telex Telex numbering plan.
[Sysname-voice] dial-program [Sysname-voice-dial] number-substitute 1 [Sysname-voice-dial-substitute1] dot-match right-left [Sysname-voice-dial-substitute1] rule 0 ^..10...$ ..267410.. If you dial the number 9810765, the number that matches the input template is 8765, and the output number is 8726741065. Examples # Configure a number substitution rule for number substitution rule list 1. The input template is ^..01...$, and the output template is ...1.
system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] send-number all Related commands match-template subscriber-group Use subscriber-group to create a subscriber group and enter subscriber group view. Use undo subscriber-group to delete a subscriber group or all subscriber groups. Syntax subscriber-group group-id undo subscriber-group { group-id | all } Default No subscriber group is created.
Default No number substitution rule list is applied to a voice entity (the voice entity does not perform number substitution). Views POTS entity view, VoIP entity view, voice interface view Predefined user roles network-admin Parameters called: Applies the number substitution rule to called numbers. calling: Applies the number substitution rule to calling numbers. list-number: Specifies a number substitution rule list by its number in the range of 1 to 2147483647.
Predefined user roles network-admin Parameters incoming-call: Applies the number substitution rule list to incoming calls. outgoing-call: Applies the number substitution rule list to outgoing calls. called: Applies the number substitution rule to called numbers. calling: Applies the number substitution rule to calling numbers. all: Specifies all number substitution rule lists. list-number: Specifies a number substitution rule list by its number in the range of 1 to 2147483647.
Predefined user roles network-admin Parameters character: Specifies a dial terminator, which is a single character that can be one of the following: • A digit from 0 through 9. • A pound sign (#). • An asterisk (*). Examples # Specify the pound sign (#) as the dial terminator.
SIP commands address sip Use address sip to configure a call destination IP address for a VoIP entity. Use undo address sip to remove the call destination IP address for a VoIP entity. Syntax address sip { dns domain-name port port-number | ip ip-address [ port port-number ]| proxy } undo address sip { dns | ip | proxy } Default No call destination IP address is configured for a VoIP entity.
Syntax asserted-id { pai | ppi } undo asserted-id Default SIP messages do not carry the P-Asserted-Identity or P-Preferred-Identity header field. Views SIP view Predefined user roles network-admin Parameters pai: Adds the P-Asserted-Identity header field into SIP messages. ppi: Adds the P-Preferred-Identity header field into SIP messages. Examples # Add the P-Asserted-Identity header field into SIP messages.
Usage guidelines You can configure source interface binding both globally (by using the bind command in SIP view) and for a specific VoIP entity (by using the voice-class sip bind command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when source interface binding is not configured in VoIP entity view.
Views SIP view Predefined user roles network-admin Parameters ssl-client-policy client-policy-name: Specifies an SSL client policy by its name, a case-insensitive string of 1 to 31 characters. ssl-server-policy server-policy-name: Specifies an SSL server policy by its name, a case-insensitive string of 1 to 31 characters.
Call 1 Call ID: 2856599de8c8824524de623ac7b1755e@200.1.1.36 Call status: Connected Calling number: 77201 Called number: 30 Control block ID: 8 Local IP address: 200.1.1.36: 5060 Remote IP address: 200.1.1.30: 5060 Media stream Media status: Send and receive Negotiated codec: g729r8 Codec payload type: 18 Codec payload size: 30 Codec transparent: Disabled Media mode: Flow-through Negotiated DTMF-relay: Inband-voice Local IP address: 200.1.1.36: 16316 Remote IP address: 200.1.1.
Field Description Media status: Media status • • • • • Send and receive. Send only. Receive only. Inactive. None. Number of SIP UAC calls Number of SIP calls initiated by the device that acts as a UAC. Number of SIP UAS calls Number of SIP calls initiated by the device that acts as a UAS. display voice sip connection Use display voice sip connection to display information about SIP connections, including established connections and connections that are being established.
Table 22 Command output Field Description Conn-Id Connection ID. Connection state: Conn-State • Connecting. • Established. Related commands reset voice sip connection ip qos dscp Use ip qos dscp to configure the global DSCP value for IP packets carrying media streams or signaling. Use undo ip qos dscp to restore the default.
Keyword DSCP value in binary DSCP value in decimal af32 011100 28 af33 011110 30 af41 100010 34 af42 100100 36 af43 100110 38 cs1 001000 8 cs2 010000 16 cs3 011000 24 cs4 100000 32 cs5 101000 40 cs6 110000 48 cs7 111000 56 ef 101110 46 Usage guidelines You can configure the ip qos dscp command both globally (in SIP view) and for a specific POTS/VoIP entity (in POTS/VoIP entity view).
Parameters pstn-sip: Displays PSTN cause-to-SIP status mappings. sip-pstn: Displays SIP status-to-PSTN cause mappings. Examples # Display PSTN cause-to-SIP status mappings.
Table 24 Command output Field Description PSTN-Cause PSTN cause code. SIP status code. SIP-Status If the configured SIP status code is different from the default, it is highlighted with an asterisk. Default Default SIP status code. # Display SIP status-to-PSTN cause mappings.
35 603 21 21 36 604 1 1 37 606 58 58 Table 25 Command output Field Description SIP-Status SIP status code. PSTN cause code. PSTN-Cause If the configured PSTN cause code is different from the default, it is highlighted with an asterisk. Default Default PSTN cause code. display voice sip register-status Use display voice sip register-status to display SIP UA registration status information.
Field Description State of the phone number: Status • • • • • • Offline. Online. Login. Logout. DNS-in—DNS query is being performed before the number is registered. DNS-out—DNS query is being performed before the number is deregistered. ip Use ip to specify a trusted node. Use undo ip to delete a trusted node. Syntax ip ipv4-address [ mask ] undo ip ipv4-address [ mask ] Default No trusted node is specified.
Default The trusted node list is disabled. Views SIP view Predefined user roles network-admin Examples # Enable the trusted node list and enter trusted node list view system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] ip address trusted list [Sysname-voice-sip-iptrust-list] min-se Use min-se to set the maximum and minimum session expiration timers. Use undo min-se to restore the default.
[Sysname-voice] sip [Sysname-voice-sip] min-se 1000 session-expires 2000 Related commands session refresh outband sip Use outband sip to enable out-of-band DTMF. Use undo outband sip to restore the default. Syntax outband sip undo outband Default Inband DTMF is enabled. Views POTS entity view, VoIP entity view Predefined user roles network-admin Usage guidelines If you use out-of-band DTMF, configure the outband sip command on both the calling and called devices.
Predefined user roles network-admin Examples # Add the Privacy header field to INVITE requests. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] privacy proxy Use proxy to configure proxy server information for a SIP UA. Use undo proxy to remove the proxy server information for a SIP UA. Syntax proxy { dns domain-name port port-number | ip ip-address [ port port-number ] } undo proxy { dns | ip } Default No proxy server information is configured for a SIP UA.
register-number Use register-number to configure a POTS entity to register the phone number with the registrar. Use undo register-number to configure a POTS entity to deregister the phone number with the registrar. Syntax register-number undo register-number Default After you complete the SIP registration configuration, a POTS entity registers the phone number with the registrar.
Predefined user roles network-admin Parameters registrar-index: Specifies an index for the registrar, in the range of 1 to 6. dns domain-name: Specifies the registrar by its domain name, which consists of case-insensitive character strings separated by dots (for example, aabbcc.com). Each separated string contains no more than 63 characters. A domain name can include letters, digits, hyphens (-), and underscores (_), and has a maximum length of 255 characters.
• display voice sip register-status • transport rel1xx Use rel1xx to configure reliable provisional responses. Use undo rel1xx to restore the default. Syntax rel1xx { disable | require value | supported value } undo rel1xx Default SIP messages carry the Supported: value header field (the rel1xx supported 100rel command applies). Views SIP view Predefined user roles network-admin Parameters disable: Disables reliable provisional responses.
Default INVITE requests includes the Remote-Party-ID header field. Views SIP view Predefined user roles network-admin Examples # Add the Remote-Party-ID header field to INVITE requests. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] remote-party-id reset voice sip connection Use reset voice sip connection to disconnect a specific SIP connection, either an established connection or a connection that is being established.
undo session refresh Default A UAC does not perform SIP session refresh. A UAS supports SIP session refresh. Views SIP view Predefined user roles network-admin Usage guidelines Use this command on a UAC. Examples # Enable SIP session refresh. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] session refresh Related commands min-se session transport Use session transport to configure the transport protocol for outgoing SIP calls.
Usage guidelines You can configure the transport protocol both globally (in SIP view) and for a specific VoIP entity (in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration. A VoIP entity uses the global configuration only when no transport protocol is configured in VoIP entity view. Configure the same transport protocol on the called and calling devices.
PSTN cause code PSTN cause description SIP status code SIP status description 22 Number changed! 410 Gone. 23 Redirection to new destination! 410 Gone. 25 Exchange routing error! 500 Server internal error. 26 Non-selected user clearing! 404 Not Found. 27 Destination out of order! 502 Bad Gateway. 28 Invalid number format (address incomplete)! 484 Address incomplete. 29 Facility rejected! 501 Not implemented. 31 Normal, unspecified! 480 Temporarily unavailable.
Views SIP view Predefined user roles network-admin Parameters pstn-code: Specifies a PSTN cause code in Table 27. The PSTN cause code 16 is invalid. sip-code: Specifies a SIP status code in Table 27. Examples # Map the PSTN cause code 17 to the SIP status code 408. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] set pstn-cause 17 sip-status 408 set sip-status Use set sip-status to configure a SIP status-to-PSTN cause mapping.
SIP status code SIP status description PSTN cause code PSTN cause description 415 Unsupported media type. 79 Service or option not implemented, unspecified! 416 Unsupported URI Scheme. 127 Interworking, unspecified! 420 Bad extension. 127 Interworking, unspecified! 421 Extension Required. 127 Interworking, unspecified! 423 Interval Too Brief. 127 Interworking, unspecified! 480 Temporarily unavailable. 18 No user responding! 481 Call/Transaction Does not Exist.
pstn-code: Specifies a PSTN cause code in Table 28. Examples # Map the SIP status code 486 to the PSTN cause code 18. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] set sip-status 486 pstn-cause 18 sip Use sip to enter SIP view. Use undo sip to remove the settings from SIP view. Syntax sip undo sip Views Voice view Predefined user roles network-admin Examples # Enter SIP view.
Parameters fallback: Supports fallback to RTP if the peer does not support SRTP. Usage guidelines The differences between the srtp and srtp fallback commands are as follows: • With the srtp command configured: { { • The device includes crypto and RTP/SAVP parameters in outgoing INVITE requests and disconnects the call after receiving a 488 response. The device can accept only calls using SRTP.
Examples # Specify the aging time as 6 minutes for TCP connections and 60 minutes for TLS connections. system-view [Sysname] voice-setup [Sysname-voice] sip [Sysname-voice-sip] timers connection aging tcp 6 [Sysname-voice-sip] timers connection aging tls 60 timers options Use timers options to set the timers options interval. Use undo timers options to restore the default. Syntax timers options value undo timers options Default The timers options interval is 500 milliseconds.
Syntax transport { tcp [ tls ] | udp } undo transport { tcp [ tls ] | udp } Default The UDP and TCP listening ports are enabled. The TLS listening port is disabled. Views SIP view Predefined user roles network-admin Parameters udp: Enables the UDP listening port (port 5060). tcp: Enables the TCP listening port (port 5060). tls: Enables the TLS listening port (port 5061). Usage guidelines You can use this command multiple times to enable multiple listening ports.
Default The SIP scheme is used globally. Views SIP view Predefined user roles network-admin Parameters sip: Specifies the SIP scheme. sips: Specifies the SIPS scheme. Usage guidelines You can configure the URL scheme both globally (by using the url command in SIP view) and for a specific VoIP entity (by using the voice-class sip url command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration.
simple: Specifies a plaintext password. password: Specifies a case-sensitive password. If simple is specified, the password must be a string of 1 to 16 characters. If cipher is specified, the password must be a string of 1 to 53 characters. realm realm: Specifies a realm, which is a string of 1 to 50 case-sensitive characters. If no realm is specified, the credentials can be used to respond to any registrars.
undo voice-class sip bind { control | media } Default The default global source interface is used. Views VoIP entity view Predefined user roles network-admin Parameters control: Specifies outbound SIP messages. media: Specifies outbound media packets. source interface interface-type interface-number: Specifies a source interface whose IP address is used as the source address of outbound SIP messages or media packets. The specified interface must be a Layer 3 Ethernet interface or dialer interface.
Views VoIP entity view Predefined user roles network-admin Parameters up-interval seconds: Sets the interval for sending OPTIONS packets when the VoIP entity is available, in the range of 5 to 1200 seconds. The default value is 60 seconds. down-interval seconds: Sets the interval for sending OPTIONS packets when the VoIP entity is not available, in the range of 5 to 1200 seconds. The default value is 30 seconds. retry retries: Sets the number of retries to change the state for the VoIP entity.
Default The default global URL scheme (SIP scheme) is used. Views VoIP entity view Predefined user roles network-admin Parameters sip: Specifies the SIP scheme. sips: Specifies the SIPS scheme. Usage guidelines You can configure the URL scheme both globally (by using the url command in SIP view) and for a specific VoIP entity (by using the voice-class sip url command in VoIP entity view). The configuration in VoIP entity view takes precedence over the global configuration.
SIP trunk commands allow-connections sip to sip Use allow-connections sip to sip to enable SIP-to-SIP calling. Use undo allow-connections sip to sip to disable SIP-to-SIP calling. Syntax allow-connections sip to sip undo allow-connections sip to sip Default SIP-to-SIP calling is disabled. Views Voice view Predefined user roles network-admin Usage guidelines After you enable SIP-to-SIP calling, the device works as a SIP trunk device. HP does not recommend using a SIP trunk device as a SIP UA.
Predefined user roles network-admin Usage guidelines If the SIP trunk device does not support any codecs on the calling and called parties, you can enable codec transparent transmission. The SIP trunk device transparently forwards codec capability sets between the two parties without intervening codec negotiation. Examples # Enable codec transparent transmission for VoIP entity 1.
matching credentials. You can configure up to 12 realms for a phone number, and up to 128 SIP trunk accounts on the device. Examples # Configure a SIP trunk account for phone number 1000 that uses the following: • Username 1000 and password 1000, for realm server1. • Username 2000 and password 2000 for realm server2. • Username 3000 and password 3000 for realm server3.
[Sysname-voice-dial] entity 1 voip [Sysname-voice-dial-entity1] media flow-around voice-class sip early-offer forced Use voice-class sip early-offer forced to enable delayed offer to early offer (DO-EO) conversion. Use undo voice-class sip early-offer forced to restore the default. Syntax voice-class sip early-offer forced undo voice-class sip early-offer forced Default DO-EO conversion is disabled.
Call services commands call-forwarding Use call-forwarding to configure call forwarding. Use undo call-forwarding to restore the default. Syntax call-forwarding { on-busy | no-reply | unavailable | unconditional } number number undo call-forwarding { on-busy | no-reply | unavailable | unconditional } Default Call forwarding is disabled. Views POTS voice entity view Predefined user roles network-admin Parameters on-busy: Enables call forwarding busy. no-reply: Enables call forwarding no reply.
system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] call-forwarding on-busy number 12345678 # Enable call forwarding unavailable and set the forwarded-to number to 12345678.
display voice mwi Use display voice mwi to display MWI information for phone numbers with subscription. Syntax display voice mwi { all | number number } Views Any view Predefined user roles network-admin network-operator mdc-operator Parameters all: Specifies all phone numbers with subscription. number number: Specifies a phone number with subscription. Examples # Display MWI information for all phone numbers with subscription.
Field Description Total number of normal messages (total number of urgent messages). Total As shown in the above example, Total: 4(3) indicates that there are 4 normal messages and 3 urgent messages in the mailbox. display voice sip subscribe-state Use display voice sip subscribe-state to display subscription information for phone numbers.
Syntax mwi undo mwi Default MWI is disabled. Views FXS interface view Predefined user roles network-admin Usage guidelines A voice entity bound to a voice interface can send SUBSCRIBE messages only after you enable MWI for the voice interface. Examples # Enable MWI for FXS interface 2/1/1. system-view [Sysname] subscriber-line 2/1/1 [Sysname-subscriber-line2/1/1] mwi mwi-server Use mwi-server to configure the voice mail server information.
expires seconds: Specifies the expiration time of the subscription, in the range of 10 to 72000 seconds. The default is 3600 seconds. transport: Specifies the transport protocol used for sending SUBSCRIBE messages. tcp: Specifies TCP as the transport protocol for sending SUBSCRIBE messages. By default, UDP is used as the transport protocol. tls: Specifies TLS as the transport protocol for sending SUBSCRIBE messages. udp: Specifies UDP as the transport protocol. By default, UDP is used.
Fax over IP commands fax cng-switch enable Use fax cng-switch enable to enable the CNG fax switchover function for a voice entity. Use undo fax cng-switch enable to restore the default. Syntax fax cng-switch enable undo fax cng-switch enable Default The CNG fax switchover function is disabled. Views POTS entity view, VoIP entity view Predefined user roles network-admin Examples # Enable the CNG fax switchover function for POTS entity 100.
Usage guidelines To use ECM, make sure the calling and called fax machines support ECM. Use the fax ecm command on both calling and called devices. Examples # Enable ECM for POTS entity 4. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 4 pots [Sysname-voice-dial-entity4] fax ecm fax level Use fax level to configure the transmit energy level. Use undo fax level to restore the default.
Use undo fax local-train threshold to restore the default. Syntax fax local-train threshold threshold undo fax local-train threshold Default The local training threshold is 10. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters threshold: Specifies the local training threshold in percentage, in the range of 0 to 100. Usage guidelines If the threshold value is exceeded during rate training, the rate training failed.
Parameters nsf value: Specifies an NSF code in hexadecimal format (2-digit country code plus 4-digit manufacturer code), in the range of 0 to 0xFFFFFF. The country code must be T.35-compliant. The value 000000 specifies standard capabilities negotiation. Examples # Configure an NSF code of 264834 for nonstandard capabilities negotiation for POTS entity 10.
Examples # Configure the standard T.38 protocol, and set the number of redundant packets to be sent for low-speed transmission to 4. system-view [Sysname] voice-setup [Sysname-voice] dial-program [Sysname-voice-dial] entity 10 pots [Sysname-voice-dial-entity10] fax protocol standard-t38 ls-redundancy 4 # Enable fax pass-through and specify the G711alaw codec for fax pass-through.
• If G.726 is used, the maximum fax rate is 14400 bps, and the corresponding modem standard is V.17. • If G.729 is used, the maximum fax rate is to 7200 bps, and the corresponding modem standard is V.29. Usage guidelines If the rate is set to a value other than disable and voice, the maximum rate is first used during rate training. If the negotiation fails, the next lower supported rate is used. Examples # Configure the maximum fax rate for rate training as 9600 bps for POTS entity 4.
modem passthrough Use modem passthrough to configure a codec and a switching mode for modem pass-through. Use undo modem passthrough to restore the default. Syntax modem passthrough { nse [ payload-type number ] | protocol } codec { g711alaw | g711ulaw } undo modem passthrough Default Modem pass-through is not used. Views POTS entity view, VoIP entity view Predefined user roles network-admin Parameters nse: Uses the named signaling event (NSE) mode to switch to modem pass-through.
Support and other resources Contacting HP For worldwide technical support information, see the HP support website: http://www.hp.
Conventions This section describes the conventions used in this documentation set. Command conventions Convention Description Boldface Bold text represents commands and keywords that you enter literally as shown. Italic Italic text represents arguments that you replace with actual values. [] Square brackets enclose syntax choices (keywords or arguments) that are optional. { x | y | ... } Braces enclose a set of required syntax choices separated by vertical bars, from which you select one.
Network topology icons Represents a generic network device, such as a router, switch, or firewall. Represents a routing-capable device, such as a router or Layer 3 switch. Represents a generic switch, such as a Layer 2 or Layer 3 switch, or a router that supports Layer 2 forwarding and other Layer 2 features. Represents an access controller, a unified wired-WLAN module, or the switching engine on a unified wired-WLAN switch. Represents an access point. Represents a mesh access point.
Index ABCDEFGHILMNOPRSTUVW A crypto,115 address sip,113 D allow-connections sip to sip,146 default,14 ani,47 delay hold,15 ani-digit,48 delay rising,15 answer enable,48 delay send-dtmf,16 area,1 delay send-wink,17 asserted-id,113 delay start-dial,17 B delay wink-hold,18 delay wink-rising,19 bind,114 description,95 busytone-detect auto,1 description,19 busytone-detect custom,2 description,77 busytone-detect period,3 dial-prefix,96 busytone-hookon delay-timer,4 dial-program,97 C d
echo-canceler enable,28 N echo-canceler tail-length,28 nlp-on,32 entity,83 number-match,102 entity hunt,98 number-substitute,103 F O fax cng-switch enable,156 open-trunk,33 fax ecm,156 outband nte,87 fax level,157 outband sip,126 fax local-train threshold,157 P fax nsf,158 passthrough,34 fax protocol,159 pcm,59 fax rate,160 plc-mode,34 fax train-mode,161 priority,103 final-callednum enable,55 privacy,126 first-rule,99 private-line,104 G proxy,127 group-b enable,56 R H re-a
timer register-pulse,71 shutdown,38 shutdown,89 timer ring-back,44 signal,39 timer wait-digit,45 silence-detect threshold,39 timers connection aging,138 sip,137 timers options,139 slic-gain,40 timeslot-set,72 special-character,65 transmit gain,45 srtp,137 transport,139 subscriber-group,109 trunk-direction,73 subscriber-line,41 ts,74 subscriber-line,66 type,46 Subscription service,163 U substitute (dial program view),110 url,140 substitute (voice entity view, voice interface view),10
